1. Field of the Invention
The present general inventive concept relates to audio encoding and decoding apparatuses and methods, and more particularly, to adaptive time/frequency-based audio encoding and decoding apparatuses and methods which can obtain high compression efficiency by making efficient use of encoding gains of two encoding methods in which a frequency-domain transform is performed on input audio data such that time-based encoding is performed on a band of the audio data suitable for voice compression and frequency-based encoding is performed on remaining bands of the audio data.
2. Description of the Related Art
Conventional voice/music compression algorithms can be broadly classified into audio codec algorithms and voice codec algorithms. Audio codec algorithms, such as aacPlus, compress a frequency-domain signal and apply a psychoacoustic model. Assuming that the audio codec and the voice codec compress voice signals have an equal amount of data, the audio codec algorithm outputs sound having a significantly lower quality than the voice codec algorithm. In particular, the quality of sound output from the audio codec algorithm is more adversely affected by an attack signal.
Voice codec algorithms, such as an adaptive multi-rate wideband codec (AMR-WB), compress a time-domain signal and apply a voicing model. Assuming that the voice codec and the audio codec compress audio signals having an equal amount of data, the voice codec algorithm outputs sound having a significantly lower quality than the audio codec algorithm.
An AMR-WB plus algorithm considers the above characteristics of the conventional voice/music compression algorithm to efficiently perform voice/music compression. In the AMR-WB plus algorithm, an algebraic code excited linear prediction (ACELP) algorithm is used as a voice compression algorithm and a Tex character translation (TCX) algorithm is used as an audio compression algorithm. In particular, the AMR-WB plus algorithm determines whether to apply the ACELP algorithm or the TCX algorithm to each processing unit, for example, each frame on a time axis, and then performs encoding accordingly. In this case, the AMR-WB plus algorithm is effective in compressing what is close to a voice signal. However, when the AMR-WB plus algorithm is used to compress what is close to an audio signal, the sound quality or compression rate deteriorates since the AMR-WB plus algorithm performs encoding in processing units.